webrtc/modules/audio_processing/gain_control_impl.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "webrtc/modules/audio_processing/gain_control_impl.h"

#include <assert.h>

#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"

namespace webrtc {

typedef void Handle;

namespace {
int16_t MapSetting(GainControl::Mode mode) {
  switch (mode) {
    case GainControl::kAdaptiveAnalog:
      return kAgcModeAdaptiveAnalog;
    case GainControl::kAdaptiveDigital:
      return kAgcModeAdaptiveDigital;
    case GainControl::kFixedDigital:
      return kAgcModeFixedDigital;
  }
  assert(false);
  return -1;
}

// Maximum length that a frame of samples can have.
static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
// Maximum number of frames to buffer in the render queue.
// TODO(peah): Decrease this once we properly handle hugely unbalanced
// reverse and forward call numbers.
static const size_t kMaxNumFramesToBuffer = 100;

}  // namespace

GainControlImpl::GainControlImpl(const AudioProcessing* apm,
                                 rtc::CriticalSection* crit_render,
                                 rtc::CriticalSection* crit_capture)
    : ProcessingComponent(),
      apm_(apm),
      crit_render_(crit_render),
      crit_capture_(crit_capture),
      mode_(kAdaptiveAnalog),
      minimum_capture_level_(0),
      maximum_capture_level_(255),
      limiter_enabled_(true),
      target_level_dbfs_(3),
      compression_gain_db_(9),
      analog_capture_level_(0),
      was_analog_level_set_(false),
      stream_is_saturated_(false),
      render_queue_element_max_size_(0) {
  RTC_DCHECK(apm);
  RTC_DCHECK(crit_render);
  RTC_DCHECK(crit_capture);
}

GainControlImpl::~GainControlImpl() {}

int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
  rtc::CritScope cs(crit_render_);
  if (!is_component_enabled()) {
    return AudioProcessing::kNoError;
  }

  assert(audio->num_frames_per_band() <= 160);

  render_queue_buffer_.resize(0);
  for (int i = 0; i < num_handles(); i++) {
    Handle* my_handle = static_cast<Handle*>(handle(i));
    int err =
        WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());

    if (err != AudioProcessing::kNoError)
      return GetHandleError(my_handle);

    // Buffer the samples in the render queue.
    render_queue_buffer_.insert(
        render_queue_buffer_.end(), audio->mixed_low_pass_data(),
        (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
  }

  // Insert the samples into the queue.
  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
    // The data queue is full and needs to be emptied.
    ReadQueuedRenderData();

    // Retry the insert (should always work).
    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
  }

  return AudioProcessing::kNoError;
}

// Read chunks of data that were received and queued on the render side from
// a queue. All the data chunks are buffered into the farend signal of the AGC.
void GainControlImpl::ReadQueuedRenderData() {
  rtc::CritScope cs(crit_capture_);

  if (!is_component_enabled()) {
    return;
  }

  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
    int buffer_index = 0;
    const int num_frames_per_band =
        capture_queue_buffer_.size() / num_handles();
    for (int i = 0; i < num_handles(); i++) {
      Handle* my_handle = static_cast<Handle*>(handle(i));
      WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
                          num_frames_per_band);

      buffer_index += num_frames_per_band;
    }
  }
}

int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
  rtc::CritScope cs(crit_capture_);

  if (!is_component_enabled()) {
    return AudioProcessing::kNoError;
  }

  assert(audio->num_frames_per_band() <= 160);
  assert(audio->num_channels() == num_handles());

  int err = AudioProcessing::kNoError;

  if (mode_ == kAdaptiveAnalog) {
    capture_levels_.assign(num_handles(), analog_capture_level_);
    for (int i = 0; i < num_handles(); i++) {
      Handle* my_handle = static_cast<Handle*>(handle(i));
      err = WebRtcAgc_AddMic(
          my_handle,
          audio->split_bands(i),
          audio->num_bands(),
          audio->num_frames_per_band());

      if (err != AudioProcessing::kNoError) {
        return GetHandleError(my_handle);
      }
    }
  } else if (mode_ == kAdaptiveDigital) {

    for (int i = 0; i < num_handles(); i++) {
      Handle* my_handle = static_cast<Handle*>(handle(i));
      int32_t capture_level_out = 0;

      err = WebRtcAgc_VirtualMic(
          my_handle,
          audio->split_bands(i),
          audio->num_bands(),
          audio->num_frames_per_band(),
          analog_capture_level_,
          &capture_level_out);

      capture_levels_[i] = capture_level_out;

      if (err != AudioProcessing::kNoError) {
        return GetHandleError(my_handle);
      }

    }
  }

  return AudioProcessing::kNoError;
}

int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
  rtc::CritScope cs(crit_capture_);

  if (!is_component_enabled()) {
    return AudioProcessing::kNoError;
  }

  if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
    return AudioProcessing::kStreamParameterNotSetError;
  }

  assert(audio->num_frames_per_band() <= 160);
  assert(audio->num_channels() == num_handles());

  stream_is_saturated_ = false;
  for (int i = 0; i < num_handles(); i++) {
    Handle* my_handle = static_cast<Handle*>(handle(i));
    int32_t capture_level_out = 0;
    uint8_t saturation_warning = 0;

    // The call to stream_has_echo() is ok from a deadlock perspective
    // as the capture lock is allready held.
    int err = WebRtcAgc_Process(
        my_handle,
        audio->split_bands_const(i),
        audio->num_bands(),
        audio->num_frames_per_band(),
        audio->split_bands(i),
        capture_levels_[i],
        &capture_level_out,
        apm_->echo_cancellation()->stream_has_echo(),
        &saturation_warning);

    if (err != AudioProcessing::kNoError) {
      return GetHandleError(my_handle);
    }

    capture_levels_[i] = capture_level_out;
    if (saturation_warning == 1) {
      stream_is_saturated_ = true;
    }
  }

  if (mode_ == kAdaptiveAnalog) {
    // Take the analog level to be the average across the handles.
    analog_capture_level_ = 0;
    for (int i = 0; i < num_handles(); i++) {
      analog_capture_level_ += capture_levels_[i];
    }

    analog_capture_level_ /= num_handles();
  }

  was_analog_level_set_ = false;
  return AudioProcessing::kNoError;
}

// TODO(ajm): ensure this is called under kAdaptiveAnalog.
int GainControlImpl::set_stream_analog_level(int level) {
  rtc::CritScope cs(crit_capture_);

  was_analog_level_set_ = true;
  if (level < minimum_capture_level_ || level > maximum_capture_level_) {
    return AudioProcessing::kBadParameterError;
  }
  analog_capture_level_ = level;

  return AudioProcessing::kNoError;
}

int GainControlImpl::stream_analog_level() {
  rtc::CritScope cs(crit_capture_);
  // TODO(ajm): enable this assertion?
  //assert(mode_ == kAdaptiveAnalog);

  return analog_capture_level_;
}

int GainControlImpl::Enable(bool enable) {
  rtc::CritScope cs_render(crit_render_);
  rtc::CritScope cs_capture(crit_capture_);
  return EnableComponent(enable);
}

bool GainControlImpl::is_enabled() const {
  rtc::CritScope cs(crit_capture_);
  return is_component_enabled();
}

int GainControlImpl::set_mode(Mode mode) {
  rtc::CritScope cs_render(crit_render_);
  rtc::CritScope cs_capture(crit_capture_);
  if (MapSetting(mode) == -1) {
    return AudioProcessing::kBadParameterError;
  }

  mode_ = mode;
  return Initialize();
}

GainControl::Mode GainControlImpl::mode() const {
  rtc::CritScope cs(crit_capture_);
  return mode_;
}

int GainControlImpl::set_analog_level_limits(int minimum,
                                             int maximum) {
  rtc::CritScope cs(crit_capture_);
  if (minimum < 0) {
    return AudioProcessing::kBadParameterError;
  }

  if (maximum > 65535) {
    return AudioProcessing::kBadParameterError;
  }

  if (maximum < minimum) {
    return AudioProcessing::kBadParameterError;
  }

  minimum_capture_level_ = minimum;
  maximum_capture_level_ = maximum;

  return Initialize();
}

int GainControlImpl::analog_level_minimum() const {
  rtc::CritScope cs(crit_capture_);
  return minimum_capture_level_;
}

int GainControlImpl::analog_level_maximum() const {
  rtc::CritScope cs(crit_capture_);
  return maximum_capture_level_;
}

bool GainControlImpl::stream_is_saturated() const {
  rtc::CritScope cs(crit_capture_);
  return stream_is_saturated_;
}

int GainControlImpl::set_target_level_dbfs(int level) {
  rtc::CritScope cs(crit_capture_);
  if (level > 31 || level < 0) {
    return AudioProcessing::kBadParameterError;
  }

  target_level_dbfs_ = level;
  return Configure();
}

int GainControlImpl::target_level_dbfs() const {
  rtc::CritScope cs(crit_capture_);
  return target_level_dbfs_;
}

int GainControlImpl::set_compression_gain_db(int gain) {
  rtc::CritScope cs(crit_capture_);
  if (gain < 0 || gain > 90) {
    return AudioProcessing::kBadParameterError;
  }

  compression_gain_db_ = gain;
  return Configure();
}

int GainControlImpl::compression_gain_db() const {
  rtc::CritScope cs(crit_capture_);
  return compression_gain_db_;
}

int GainControlImpl::enable_limiter(bool enable) {
  rtc::CritScope cs(crit_capture_);
  limiter_enabled_ = enable;
  return Configure();
}

bool GainControlImpl::is_limiter_enabled() const {
  rtc::CritScope cs(crit_capture_);
  return limiter_enabled_;
}

int GainControlImpl::Initialize() {
  int err = ProcessingComponent::Initialize();
  if (err != AudioProcessing::kNoError || !is_component_enabled()) {
    return err;
  }

  AllocateRenderQueue();

  rtc::CritScope cs_capture(crit_capture_);
  const int n = num_handles();
  RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;

  capture_levels_.assign(n, analog_capture_level_);
  return AudioProcessing::kNoError;
}

void GainControlImpl::AllocateRenderQueue() {
  const size_t new_render_queue_element_max_size =
      std::max<size_t>(static_cast<size_t>(1),
                       kMaxAllowedValuesOfSamplesPerFrame * num_handles());

  rtc::CritScope cs_render(crit_render_);
  rtc::CritScope cs_capture(crit_capture_);

  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
    render_queue_element_max_size_ = new_render_queue_element_max_size;
    std::vector<int16_t> template_queue_element(render_queue_element_max_size_);

    render_signal_queue_.reset(
        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
            kMaxNumFramesToBuffer, template_queue_element,
            RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));

    render_queue_buffer_.resize(render_queue_element_max_size_);
    capture_queue_buffer_.resize(render_queue_element_max_size_);
  } else {
    render_signal_queue_->Clear();
  }
}

void* GainControlImpl::CreateHandle() const {
  return WebRtcAgc_Create();
}

void GainControlImpl::DestroyHandle(void* handle) const {
  WebRtcAgc_Free(static_cast<Handle*>(handle));
}

int GainControlImpl::InitializeHandle(void* handle) const {
  rtc::CritScope cs_render(crit_render_);
  rtc::CritScope cs_capture(crit_capture_);

  return WebRtcAgc_Init(static_cast<Handle*>(handle),
                          minimum_capture_level_,
                          maximum_capture_level_,
                          MapSetting(mode_),
                          apm_->proc_sample_rate_hz());
}

int GainControlImpl::ConfigureHandle(void* handle) const {
  rtc::CritScope cs_render(crit_render_);
  rtc::CritScope cs_capture(crit_capture_);
  WebRtcAgcConfig config;
  // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
  //            change the interface.
  //assert(target_level_dbfs_ <= 0);
  //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
  config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
  config.compressionGaindB =
      static_cast<int16_t>(compression_gain_db_);
  config.limiterEnable = limiter_enabled_;

  return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
}

int GainControlImpl::num_handles_required() const {
  // Not locked as it only relies on APM public API which is threadsafe.
  return apm_->num_output_channels();
}

int GainControlImpl::GetHandleError(void* handle) const {
  // The AGC has no get_error() function.
  // (Despite listing errors in its interface...)
  assert(handle != NULL);
  return AudioProcessing::kUnspecifiedError;
}
}  // namespace webrtc